1. Field of the Invention
The present invention relates to a method and apparatus for matching sound quality measurement sections of a variable bandwidth multi-codec, and more particularly, a method and apparatus for matching sound quality measurement sections of a variable bandwidth multi-code, the method and apparatus capable of enabling a sound quality measurement apparatus to compare transmission rates of the variable bandwidth multi-codec according to transmission capabilities of a network, transmission delays in the network, packet losses, central processing unit (CPU) usage rates, a natural sound and an output sound of the variable bandwidth multi-codec, and the output sound of the variable bandwidth multi-codec and a sound received over the network for the same sound quality measurement sections when a real-time multimedia service is provided using the variable bandwidth multi-codec, which provides different transmission rates to a caller and the called, and using a connection function between a packet network and an existing wired/wireless network.
The present invention is supported by an information technology (IT) research and development (R&D) program of Ministry of Information and Communication (MIC)/Institute for Information Technology Advancement (IITA) [2005-S-100-02, “Development of Multi-Codec and Its Control Technology Providing Variable Bandwidth Scalability”].
2. Description of the Related Art
A variable bandwidth multi-codec is a technology for converting a natural sound into digitized codec data having a plurality of transmission rates. One example is a codec technology that can divide a frequency band into a narrowband (from 300 Hz to 3,400 Hz), a broadband (from 50H to 7,000 Hz) and an audio band (from 20 to 20,000 Hz), and produce transmission rates of 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, and 32 bps in each bandwidth.
In a voice over Internet protocol (VoIP) voice call service, a bandwidth provided by a network is variable and unpredictable. For example, a transmission rate of 32 bps of a variable bandwidth multi-codec is a codec transmission rate that produces the best sound quality, and a transmission rate of 8 kpbs is a codec transmission rate that produces the worst sound quality. If the network has available bandwidth and thus data having high bit rate can be, data can be transmitted at a transmission rate of 32 kbps. If the network condition deteriorates due to changes in the network bandwidth, the transmission rate is lowered to 30 kbps, 28 kbps, or the like, so that data can be slowly transmitted over the network although their quality is degraded. In the variable bandwidth multi-codec, if the transmission rate is high, sound quality is good. However, there is a high probability that a transmission loss and delay will occur in the network. On the other hand, if the transmission rate is low, sound quality is poor. However, there is a low probability that the transmission loss and delay will occur in the network.
When sound quality of the variable bandwidth multi-codec is measured on an end-to-end manner, it can be accurately measured, thereby making it possible to accurately control the transmission rate. That is, accurate control of the transmission rate requires accurate measurement of sound quality. Sound quality measurement refers to a process of comparing a file in which a natural sound is recorded to a file in which an output sound of a codec is recorded. In this process, if sound sections included in the two files are different from each other, accurate sound quality measurement is not possible.
Conventional technologies cannot find causes of sound quality distortion that occurs while subjects of end-to-end sound quality measurement are selected. Hence, a method and apparatus for finding causes of sound quality distortion by measuring a natural sound, a file stored after original sound data is converted using a variable bandwidth multi-codec according to changes in the condition of network bandwidth, and a group of files received over a network by using a sound quality measurement algorithm are required.